Commit Graph

425 Commits

Author SHA1 Message Date
2e3069bf07 Use ScopedFieldTrials in FieldTrialsTest
Resetting the global state between runs was previously handled by a
RAII type, but the semantics of that type changed to remove this
behavior in [1].

[1] https://webrtc-review.googlesource.com/c/src/+/276269

Bug: webrtc:14731, webrtc:14705
Change-Id: I8425cb71f49ea000434d500e0b3978324e4c3195
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285782
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38800}
2022-12-02 15:57:57 +00:00
adf35a359e Extend mocks for public types
Extends the mocks for rtpreceiver rtpsender and videotrack. This change
allows the external HangoutsKit client to remove its own mocks of rtc
types.

Bug: none
Change-Id: I8ba1752fe7633f9e0bba264a1279f74cc1368a2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jack Smith <jackdsmith@google.com>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38782}
2022-11-30 19:01:40 +00:00
5c4509a604 Add a clone method to the video frame transformer API.
This will clone an encoded video frame into a sender frame,
preserving metadata as much as possible.

Bug: webrtc:14708
Change-Id: I6f68d2ee65ef85c32cc3c142a41346b81ba73533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284701
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38733}
2022-11-25 11:18:22 +00:00
6eb1e709da Reland "[DVQA] Create separate BUILD.gn file for video analyzer"
This reverts commit 76793c300fdd87fa8fd8be3dd2e5faf8c1916e96.

Reason for revert: Can't cleanly revert the old one. A forward fix will be provided.

Original change's description:
> Revert "[DVQA] Create separate BUILD.gn file for video analyzer"
>
> This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c.
>
> Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview
>
>
> Original change's description:
> > [DVQA] Create separate BUILD.gn file for video analyzer
> >
> > Bug: None
> > Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17
> > No-try: True
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141
> > Commit-Queue: Artem Titov <titovartem@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38662}
>
> Bug: None
> Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000
> Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38672}

Bug: None
Change-Id: I74506eaa6a1060bf87e651881c86b4f576f447ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284020
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38676}
2022-11-18 11:43:45 +00:00
76793c300f Revert "[DVQA] Create separate BUILD.gn file for video analyzer"
This reverts commit 116c0a53d4a35c6dee857eb4cc2b6ae233a0427c.

Reason for revert: Breaks bot: https://ci.chromium.org/ui/p/chromium/builders/try/linux_chromium_compile_dbg_ng/1415352/overview


Original change's description:
> [DVQA] Create separate BUILD.gn file for video analyzer
>
> Bug: None
> Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17
> No-try: True
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38662}

Bug: None
Change-Id: Ieeb8c569560cb9d60d0c4d3c1268fa57f56b8157
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284000
Auto-Submit: Ilya Nikolaevskiy <ilnik@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38672}
2022-11-18 09:18:32 +00:00
116c0a53d4 [DVQA] Create separate BUILD.gn file for video analyzer
Bug: None
Change-Id: I37dd2262bf3f52b2f5abe7934b9c41eaa27ffd17
No-try: True
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283141
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38662}
2022-11-17 11:53:44 +00:00
acabb3641b pc: Add asynchronous RtpSender::SetParameters() call
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.

Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
2022-11-15 15:31:40 +00:00
1bef09708a Delete api/stats_types.h in favor of api/legacy_stats_types.h
The file was renamed, see
https://groups.google.com/u/1/g/discuss-webrtc/c/ZQiP4f_bpw4

Bug: webrtc:14180
Change-Id: Ia76c85ba7d9da6b3a93d0a67a4b6a5187e07e230
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283084
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38616}
2022-11-14 12:10:06 +00:00
3e6931b183 Rename api/stats_types.h to api/legacy_stats_types.h.
As to not break downstream projects, the old name api/stats_types.h is
kept around to help include api/legacy_stats_types.h. We can delete this
in a follow-up.

NOTRY=True

Bug: webrtc:14180
Change-Id: I270ca5e366ae36e324cbc9f982bbb066ab92d203
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283081
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38610}
2022-11-11 10:29:25 +00:00
a3e51df5f3 Add a new PeerConnectionE2EQualityTestFixture::AddPeer method.
Change-Id: Ic5879613db51a00e3e958931f5eda19fda1ae94a
Bug: webrtc:14627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282640
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38608}
2022-11-10 16:54:19 +00:00
123a0ed604 Revert "Add checks for api/test mocks to make sure they're complete"
This reverts commit e87ec28b807f84babe228f54690c686fcf86a0fb.

Reason for revert: Breaks upstream.

Original change's description:
> Add checks for api/test mocks to make sure they're complete
>
> Also unifies the mock inheritance if they inherited from a ref counted
> interface:
>  - it should only inherit from the interface
>  - it should use make_ref_counted
>
> Bug: webrtc:14594
> Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423
> Commit-Queue: Florent Castelli <orphis@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38602}

Bug: webrtc:14594
Change-Id: I9f2d9c3656b43e3006ec03ae7d792d0a53f47ebd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282940
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38604}
2022-11-10 13:33:59 +00:00
389228d0f0 Remove PeerConfigurer interface.
PeerConfigurerImpl is renamed to PeerConfigurer.

Change-Id: Ie52c581126c21740536d42ff4831f0c4ed445ea4
Bug: webrtc:14627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281883
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38603}
2022-11-10 12:52:25 +00:00
e87ec28b80 Add checks for api/test mocks to make sure they're complete
Also unifies the mock inheritance if they inherited from a ref counted
interface:
 - it should only inherit from the interface
 - it should use make_ref_counted

Bug: webrtc:14594
Change-Id: I7b0514b632ccd0798028b50f19812ac0a196e13c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/262423
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38602}
2022-11-10 12:47:31 +00:00
0e2cf6cc01 Use classes from media_configuration.h instead of the ones in PeerConnectionE2EQualityTestFixture.
Classes defined inside the class PeerConnectionE2EQualityTestFixture are replaced by the ones define in media_configuration.h.

Change-Id: I1c025ff10aacf8cbc3df9bfa742a40622fe0807a
Bug: webrtc:14627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281860
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38568}
2022-11-07 16:56:47 +00:00
e91d4bc517 Move media configuration classes out of PeerConnectionE2EQualityTestFixture.
The goal is to remove the dependency between PeerConfigurerImpl and PeerConnectionE2EQualityTestFixture so that PeerConfigurerImpl can be used in PeerConnectionE2EQualityTestFixture API.

Change-Id: I29ae44b9d0e39075d0c395ff9d9f8d313be12176
Bug: webrtc:14627
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281740
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Jeremy Leconte <jleconte@google.com>
Cr-Commit-Position: refs/heads/main@{#38560}
2022-11-07 09:34:59 +00:00
96002fa8da [PCLF] Include video resolution into video dump file name
Bug: b/240540204
Change-Id: Idad6a5c67c2dcedb07cfa915ac986590c1e29275
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280383
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#38470}
2022-10-25 17:21:47 +00:00
98fe985480 Add MockSessionDescriptionInterface
This is needed to get rid of a mock in Chrome.

Bug: webrtc:14594
Change-Id: I27df2a1466e6a2dea87a211f803b3f2c7aa57478
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280041
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38458}
2022-10-24 12:21:31 +00:00
1c8103d4db Add FieldTrialsRegistry that verifies looked up field trials
This new class implements the existing FieldTrialsView interface,
extending it with the verification functionality. For now, the
verification will only be performed if the rtc_strict_field_trials GN
arg is set.

Most classes extending FieldTrialsView today have been converted to
extend from FieldTrialsRegistry instead to automatically perform
verification.

Bug: webrtc:14154
Change-Id: I4819724cd66a04507e62fcc2bb1019187b6ba8c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276270
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38453}
2022-10-24 09:12:30 +00:00
6bf20cc76a Verify field trials looked up through field_trial::FindFullName
For now, the run-time check will only be enabled if the
rtc_strict_field_trials GN arg is set.

In order to allow testing with imaginary field trial keys, two test
helpers have been added. It's a bit awkward to test these since the
field trial string is already global, hence the helpers are also
modifying global state. Tests must make sure this global state is reset
between runs. Things won't be an issue anymore when [1] has removed the
global string.

[1] https://crbug.com/webrtc/10335

Bug: webrtc:14154
Change-Id: Ida44cc817079d7177325e2228cf1f1d242b799e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276269
Commit-Queue: Emil Lundmark <lndmrk@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38447}
2022-10-20 10:46:01 +00:00
8e7a105c51 stats: use absl::optional to represent value
which is a more modern way to represent something that either has a value or is not set

BUG=webrtc:14544

Change-Id: I0a06b30b1c7f802247eb1f60e69271594b94a6f0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278421
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38443}
2022-10-19 15:57:30 +00:00
036b3fdea2 Reland "stats: migrate to Timestamp"
This is a reland of commit 2235776597e2f47ec353ac911428eb9a54d64a10

Original change's description:
> stats: migrate to Timestamp
>
> BUG=webrtc:13756
>
> Change-Id: I04ba57f9c2ca5a974a406814023911b4eb2d6d38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273942
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38365}

Bug: webrtc:13756
Change-Id: Ib8dc208197ae5e90f67114e7b043a73ee35421ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279080
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38380}
2022-10-13 09:03:43 +00:00
c0794c23ff Revert "stats: migrate to Timestamp"
This reverts commit 2235776597e2f47ec353ac911428eb9a54d64a10.

Reason for revert: Breaks compile.

RTCStatsReport::Create(timestamp) needs default value.

Original change's description:
> stats: migrate to Timestamp
>
> BUG=webrtc:13756
>
> Change-Id: I04ba57f9c2ca5a974a406814023911b4eb2d6d38
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273942
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38365}

Bug: webrtc:13756
Change-Id: I7eba2bb510af73be50397bd92f730bc6de1ce676
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279044
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38369}
2022-10-12 14:23:40 +00:00
2068d0daa7 [PCLF] Add ability to provide custom VideoFrameWriter
Bug: b/240540204
Change-Id: Ica85954ea61b7caf4e2d726895b6a439b47d7bbb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278800
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38368}
2022-10-12 14:08:00 +00:00
2235776597 stats: migrate to Timestamp
BUG=webrtc:13756

Change-Id: I04ba57f9c2ca5a974a406814023911b4eb2d6d38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273942
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38365}
2022-10-12 11:43:39 +00:00
5c9b7da038 Add missing dependencies.
Bug: b/251890128
Change-Id: Ia9312797a5552ad1ceb4a80968014b849121a1b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278580
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Christoffer Jansson <jansson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38333}
2022-10-10 10:18:37 +00:00
9d9c2d5795 Make header files self contained.
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.

Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
2022-10-08 08:38:36 +00:00
6c2dae21e9 Move VideoEncoderConfig from api/ into video/config
This cl move VideoEncoderConfig from api/ to video/config.

VideoStreamEncoderInterface and VideoStreamEncoderObserver
are moved as collateral.

brandt@ think that the reason these were in api/ in the
first place had to downstream project.

Functionality wise, this is a NOP, but it makes it easier
to modify the encoder (config).

Bug: webrtc:14451
Change-Id: I2610d815aeb186298498e7102cac773ecac8cd36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277002
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38242}
2022-09-29 09:44:43 +00:00
0deda15c96 Reland "RtpEncodingParameters::request_resolution patch 1"
This reverts commit b625101da8d798c936cfd695505a5514644158b0.

Reason for revert: Found problem that was specific how
configuration is handled for VP9. A 1-line change in webrtc_video_engine.cc line 3715.
Thanks Rasmus and great that this was tested!

Original change's description:
> Revert "RtpEncodingParameters::request_resolution patch 1"
>
> This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2.
>
> Reason for revert: Breaks downstream test
>
> Original change's description:
> > RtpEncodingParameters::request_resolution patch 1
> >
> > This patch adds RtpEncodingParameters::request_resolution
> > with documentation and plumming. No behaviour is changed yet.
> >
> > Bug: webrtc:14451
> > Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
> > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Reviewed-by: Henrik Boström <hbos@webrtc.org>
> > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> > Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> > Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#38172}
>
> Bug: webrtc:14451
> Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
> Owners-Override: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38176}

Bug: webrtc:14451
Change-Id: Ica9b74180bce22d09bf289126bb5ac137bf9eb70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276543
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38178}
2022-09-23 11:48:19 +00:00
b625101da8 Revert "RtpEncodingParameters::request_resolution patch 1"
This reverts commit ef7359e679e579ccb79afacf5c42e8c6020124e2.

Reason for revert: Breaks downstream test

Original change's description:
> RtpEncodingParameters::request_resolution patch 1
>
> This patch adds RtpEncodingParameters::request_resolution
> with documentation and plumming. No behaviour is changed yet.
>
> Bug: webrtc:14451
> Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38172}

Bug: webrtc:14451
Change-Id: I4b9590e23ec38e9e1c2e51a4600ef96b129439f2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276541
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Owners-Override: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38176}
2022-09-23 08:27:47 +00:00
ef7359e679 RtpEncodingParameters::request_resolution patch 1
This patch adds RtpEncodingParameters::request_resolution
with documentation and plumming. No behaviour is changed yet.

Bug: webrtc:14451
Change-Id: I1f4f83a312ee8c293e3d8f02b950751e62048304
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276262
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38172}
2022-09-22 14:16:20 +00:00
3680605caa [PCLF] Enable exporting of perf metric via new API
Bug: b/246095034
Change-Id: I05f28e5dfc6df793c035110f89d9ac40783687f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276267
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38161}
2022-09-21 15:55:09 +00:00
56b96ffe6a Surface local_capture_clock_offset from RtpSource
- Propagating `RtpPacketInfo::local_capture_clock_offset`, an
  existing field that is related to the abs-capture-timestamp
  header extension field `estimated_capture_clock_offset`
- Propagated through `SourceTracker::SourceEntry`

Bug: webrtc:10739, b/246753278
Change-Id: I21d9841e4f3a35da5f8d7b31582898309421d524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275241
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38129}
2022-09-20 12:51:22 +00:00
24c1079b2f Reland "rtpsender interface: make pure virtual again"
This reverts commit fbb7ce8a935db1988b3571639cab1eaed88980d1.

Reason for revert: Relanding because the upstream project should be compatible with the changes now.

Original change's description:
> Revert "rtpsender interface: make pure virtual again"
>
> This reverts commit 021512b76a872b04e803d61f46c740ed363d641b.
>
> Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly.
>
> Original change's description:
> > rtpsender interface: make pure virtual again
> >
> > after providing default implementations in Chromium tests
> >
> > BUG=None
> >
> > Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
> > Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> > Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37941}
>
> Bug: None
> Change-Id: I40f27c36819365fadae32032521f7e11184bee62
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484
> Owners-Override: Andrey Logvin <landrey@google.com>
> Commit-Queue: Andrey Logvin <landrey@google.com>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Andrey Logvin <landrey@google.com>
> Cr-Commit-Position: refs/heads/main@{#37947}

Bug: None
Change-Id: I531e17d5252d4bd5450d5ac5c64fc8f51b4a1d1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273701
Commit-Queue: Andrey Logvin <landrey@google.com>
Reviewed-by: Philipp Hancke <phancke@microsoft.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Owners-Override: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37969}
2022-08-31 14:47:14 +00:00
fbb7ce8a93 Revert "rtpsender interface: make pure virtual again"
This reverts commit 021512b76a872b04e803d61f46c740ed363d641b.

Reason for revert: Breaks upstream project. It relies on the default implementation. The CL will be relanded after the migration is done. We will make sure to do it shortly.

Original change's description:
> rtpsender interface: make pure virtual again
>
> after providing default implementations in Chromium tests
>
> BUG=None
>
> Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
> Commit-Queue: Philipp Hancke <phancke@microsoft.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37941}

Bug: None
Change-Id: I40f27c36819365fadae32032521f7e11184bee62
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273484
Owners-Override: Andrey Logvin <landrey@google.com>
Commit-Queue: Andrey Logvin <landrey@google.com>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Andrey Logvin <landrey@google.com>
Cr-Commit-Position: refs/heads/main@{#37947}
2022-08-30 11:27:50 +00:00
021512b76a rtpsender interface: make pure virtual again
after providing default implementations in Chromium tests

BUG=None

Change-Id: I53bf26b3a99416f4005e7df75b9b86dfbf2489cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273100
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37941}
2022-08-30 09:19:45 +00:00
b787e26369 Support DNS resolution matching a specified IP family.
The input SocketAddress for STUN host lookup is constructed with just
the hostname, so the family is AF_UNSPEC. So added an overload with a
target family to distinguish this from the family of the input addr.

Bug: webrtc:14319, webrtc:14131
Change-Id: Ia5ac5aa2e894e0c4dfb4417e3e8a76a6cec3ea71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270624
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Sameer Vijaykar <samvi@google.com>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@google.com>
Cr-Commit-Position: refs/heads/main@{#37750}
2022-08-11 13:52:53 +00:00
72bc2e24cc [PCLF] Introduce VideoDumpOptions API to better control video dumping
This CL propose a new API for video dumps in PCLF also removing
differences between p2p and multipeer usage of API.

Bug: b/240540206
Change-Id: Id4d32cc98250500949b3f9e2cf2e86c4bdce7efb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270400
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37665}
2022-08-02 15:13:04 +00:00
a1a7c638ec Let PCF.GetRtpSenderCapabilities return codecs' scalabilityModes.
Also move ScalabilityModeToString to api and add RTC_EXPORT so that
Chromium can use it.

Bug: chromium:986069
Change-Id: I5dbbb6de9b14ca20f3ae0630552dcd44595ad5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37444}
2022-07-05 13:28:33 +00:00
63299a3124 Add absl::string_view overload for RtcEventLogOutput::Write
Bug: webrtc:13579
Change-Id: I13f63fb6be6aa62c2e011c18327499fa16b5824e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267641
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37440}
2022-07-05 10:47:47 +00:00
24b0543ee0 Delete ProcessThread creation from test TimeController as unused
Bug: webrtc:7219
Change-Id: Ia34f24a804b8a1e06b089774e37cac6e6d749e82
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266366
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37311}
2022-06-23 09:41:43 +00:00
d151cc6fa3 Remove the last build cycle in WebRTC
This CL removes the last "nogncheck" comment that was related to a
known build cycle. The remaining ones are because of conditional
dependencies.

Bug: webrtc:8733
Change-Id: Ie6862ae1cc613b9c2740a34c3167e1741ed31ee3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265981
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37302}
2022-06-22 10:44:51 +00:00
573b145ab5 Reland "Move injection of PacketSocketFactory from PC to PCF"
This is a reland of commit 905c3a6c73d293882ef11942066ccda52a9e14d1

Change from previous attempt is between ps#1 and ps#2: Use PeerConnectionFactoryInterface::Options to clear the `network_ignore_mask`.

Original change's description:
> Move injection of PacketSocketFactory from PC to PCF
>
> Injection via PeerConnectionDependecies was broken, in not accepting
> ownership of the injected object.
>
> Bug: webrtc:7447, webrtc:14204
> Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37270}

Bug: webrtc:7447, webrtc:14204
Change-Id: Ic78ebec2e88a8c44699015c8c7a44e137f44253a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265982
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37290}
2022-06-21 10:28:39 +00:00
3627d57849 Revert "Move injection of PacketSocketFactory from PC to PCF"
This reverts commit 905c3a6c73d293882ef11942066ccda52a9e14d1.

Reason for revert: New test fails internal tests, with a similar problem as the failed android test: No networks are detected on the test bot.

Original change's description:
> Move injection of PacketSocketFactory from PC to PCF
>
> Injection via PeerConnectionDependecies was broken, in not accepting
> ownership of the injected object.
>
> Bug: webrtc:7447, webrtc:14204
> Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37270}

Bug: webrtc:7447, webrtc:14204
Change-Id: Ib412d09699a48d8f5db27e2960e365b536ab3aa8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266146
Owners-Override: Niels Moller <nisse@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37273}
2022-06-20 09:50:11 +00:00
905c3a6c73 Move injection of PacketSocketFactory from PC to PCF
Injection via PeerConnectionDependecies was broken, in not accepting
ownership of the injected object.

Bug: webrtc:7447, webrtc:14204
Change-Id: Ic53f05d51928b006fc1e46d502633d88471eb518
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266140
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37270}
2022-06-20 08:32:02 +00:00
130f2fd5c3 Fix missing visibility for new API targets.
Bug: webrtc:8733
Change-Id: I3fb1ac9a9941f7b8ad35a51be0e7b6840a2d9141
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266220
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37265}
2022-06-18 13:34:40 +00:00
01ed32589e Introduce empty targets to split libjingle_peerconnection_api.
First step of the process to remove the last cycle in the build graph.

Bug: webrtc:8733
Change-Id: I5a0c987ce3d602d1cb30991b73b68a389f13cfba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265874
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37261}
2022-06-17 20:14:19 +00:00
7c8c4db8ea Add rtc::make_ref_counted to api/
This cl adds a forwarding header, a build target, and migrates headers
in api/ to use it.

Moving actual implementation, will follow, in
https://webrtc-review.googlesource.com/c/src/+/265390.

Bug: webrtc:12701
Change-Id: Id950725d7d054de8a83b3800b9c9a6437344de86
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265660
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37200}
2022-06-13 15:53:27 +00:00
10da2b5357 Delete unused test class DummyPeerConnection
Introduced in https://webrtc-review.googlesource.com/c/src/+/152820
but never used.

Bug: None
Change-Id: I8888db9e0cad39a6eb073e8f662b5ad2690d03e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265409
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37173}
2022-06-10 07:29:54 +00:00
6545516a14 RtpSenderInterface::SetEncoderSelector
This cl/ adds a way of setting an EncoderSelector on a specific
RtpSenderInterface. This makes it possible to easily use different
EncoderSelector on different streams within the same or different PeerConnections.

The cl/ is almost identical to the impl. of RtpSenderInterface::SetFrameEncryptor.

Iff a EncoderSelector is set on the RtpSender, it will take precedence
over the VideoEncoderFactory::GetEncoderSelector.

Bug: webrtc:14122
Change-Id: Ief4f7c06df7f1ef4ce3245de304a48e9de0ad587
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264542
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37150}
2022-06-08 11:06:52 +00:00
b22cabcf76 Add AbslParse functions for TimeMode enum. (This allows creation of TimeMode ABSL_FLAGs.)
Bug: webrtc:14145
Change-Id: Id79c4411ba4443a3ee8a0da3990c36955cc9aa35
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264821
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Auto-Submit: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37090}
2022-06-02 10:07:16 +00:00